[svn-commits] file: branch file/rtp_engine r129166 - in /team/file/rtp_engine: channels/ in...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jul 8 16:20:41 CDT 2008
Author: file
Date: Tue Jul 8 16:20:40 2008
New Revision: 129166
URL: http://svn.digium.com/view/asterisk?view=rev&rev=129166
Log:
Hey look! It compiles!
Modified:
team/file/rtp_engine/channels/chan_sip.c
team/file/rtp_engine/include/asterisk/rtp_engine.h
team/file/rtp_engine/res/res_rtp_asterisk.c
Modified: team/file/rtp_engine/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/file/rtp_engine/channels/chan_sip.c?view=diff&rev=129166&r1=129165&r2=129166
==============================================================================
--- team/file/rtp_engine/channels/chan_sip.c (original)
+++ team/file/rtp_engine/channels/chan_sip.c Tue Jul 8 16:20:40 2008
@@ -5931,12 +5931,18 @@
p->ocseq = INITIAL_CSEQ;
if (sip_methods[intended_method].need_rtp) {
- p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ //p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ p->rtp = ast_rtp_instance_new(NULL, NULL);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
/* If the global videosupport flag is on, we always create a RTP interface for video */
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT))
- p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
- if (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT))
- p->trtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT)) {
+ p->vrtp = ast_rtp_instance_new(NULL, NULL);
+ ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+ }
+ if (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT)) {
+ p->trtp = ast_rtp_instance_new(NULL, NULL);
+ ast_rtp_instance_set_prop(p->trtp, AST_RTP_PROPERTY_RTCP, 1);
+ }
if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT))
p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
if (!p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)
@@ -6855,14 +6861,13 @@
if (framing && p->autoframing) {
struct ast_codec_pref *pref = &p->rtp->codecs.pref;
int codec_n;
- int format = 0;
for (codec_n = 0; codec_n < AST_RTP_MAX_PT; codec_n++) {
- format = ast_rtp_codec_getformat(codec_n);
- if (!format) /* non-codec or not found */
+ struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(&p->rtp->codecs, codec_n);
+ if (!format.isAstFormat || !format.code) /* non-codec or not found */
continue;
if (option_debug)
- ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing);
- ast_codec_pref_setsize(pref, format, framing);
+ ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format.code, framing);
+ ast_codec_pref_setsize(pref, format.code, framing);
}
ast_rtp_codecs_packetization_set(&p->rtp->codecs, p->rtp, pref);
}
@@ -6875,7 +6880,7 @@
sscanf(red_cp, "%u", &red_data_pt[red_num_gen]);
red_cp = strtok(red_cp, "/");
- while (red_cp && red_num_gen++ < RED_MAX_GENERATION) {
+ while (red_cp && red_num_gen++ < AST_RED_MAX_GENERATION) {
sscanf(red_cp, "%u", &red_data_pt[red_num_gen]);
red_cp = strtok(NULL, "/");
}
@@ -8166,11 +8171,11 @@
struct sockaddr_in *dest, struct sockaddr_in *vdest)
{
/* First, get our address */
- ast_rtp_get_us(p->rtp, sin);
+ memcpy(sin, &p->rtp->remote_address, sizeof(*sin));
if (p->vrtp)
- ast_rtp_get_us(p->vrtp, vsin);
+ memcpy(vsin, &p->vrtp->remote_address, sizeof(*vsin));
if (p->trtp)
- ast_rtp_get_us(p->trtp, tsin);
+ memcpy(tsin, &p->trtp->remote_address, sizeof(*tsin));
/* Now, try to figure out where we want them to send data */
/* Is this a re-invite to move the media out, then use the original offer from caller */
@@ -13274,12 +13279,12 @@
#define FORMAT2 "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s (%-2.2s) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
#define FORMAT "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u\n"
struct sip_pvt *cur = __cur;
- unsigned int rxcount;
- unsigned int txcount;
- char durbuf[10];
- int duration;
- int durh, durm, durs;
- struct ast_channel *c = cur->owner;
+// unsigned int rxcount;
+// unsigned int txcount;
+// char durbuf[10];
+ // int duration;
+ // int durh, durm, durs;
+ //struct ast_channel *c = cur->owner;
struct __show_chan_arg *arg = __arg;
int fd = arg->fd;
@@ -15281,7 +15286,7 @@
if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
change_t38_state(p, T38_DISABLED);
/* Try to reset RTP timers */
- ast_rtp_set_rtptimers_onhold(p->rtp);
+ //ast_rtp_set_rtptimers_onhold(p->rtp);
/* Trigger a reinvite back to audio */
transmit_reinvite_with_sdp(p, FALSE, FALSE);
@@ -15290,7 +15295,7 @@
right now we can't fall back to audio so totally abort.
*/
/* Try to reset RTP timers */
- ast_rtp_set_rtptimers_onhold(p->rtp);
+ //ast_rtp_set_rtptimers_onhold(p->rtp);
ast_log(LOG_ERROR, "Got error on T.38 initial invite. Bailing out.\n");
change_t38_state(p, T38_DISABLED);
@@ -17803,7 +17808,7 @@
static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
{
struct sip_pvt *p = chan->tech_pvt;
- char *all = "", *parse = ast_strdupa(preparse);
+ char /**all = "",*/ *parse = ast_strdupa(preparse);
int res = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(param);
@@ -21742,13 +21747,6 @@
return 0;
}
-/*! \brief Return SIP UA's codec (part of the RTP interface) */
-static int sip_get_codec(struct ast_channel *chan)
-{
- struct sip_pvt *p = chan->tech_pvt;
- return p->peercapability ? p->peercapability : p->capability;
-}
-
/*! \brief Send a poke to all known peers
Space them out 100 ms apart
XXX We might have a cool algorithm for this or use random - any suggestions?
Modified: team/file/rtp_engine/include/asterisk/rtp_engine.h
URL: http://svn.digium.com/view/asterisk/team/file/rtp_engine/include/asterisk/rtp_engine.h?view=diff&rev=129166&r1=129165&r2=129166
==============================================================================
--- team/file/rtp_engine/include/asterisk/rtp_engine.h (original)
+++ team/file/rtp_engine/include/asterisk/rtp_engine.h Tue Jul 8 16:20:40 2008
@@ -29,6 +29,8 @@
/* Maximum number of payloads supported */
#define AST_RTP_MAX_PT 256
+
+#define AST_RED_MAX_GENERATION 5
struct ast_rtp_instance;
struct ast_rtp_glue;
Modified: team/file/rtp_engine/res/res_rtp_asterisk.c
URL: http://svn.digium.com/view/asterisk/team/file/rtp_engine/res/res_rtp_asterisk.c?view=diff&rev=129166&r1=129165&r2=129166
==============================================================================
--- team/file/rtp_engine/res/res_rtp_asterisk.c (original)
+++ team/file/rtp_engine/res/res_rtp_asterisk.c Tue Jul 8 16:20:40 2008
@@ -224,7 +224,7 @@
static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit);
static void ast_rtp_new_source(struct ast_rtp_instance *instance);
static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
-static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance);
+static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
/* RTP Engine Declaration */
static struct ast_rtp_engine asterisk_rtp_engine = {
@@ -723,7 +723,7 @@
return 0;
}
-static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance)
+static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp)
{
struct ast_rtp *rtp = instance->data;
struct sockaddr_in sin;
@@ -733,7 +733,9 @@
struct ast_rtp_payload_type payload;
/* If this is actually RTCP let's hop on over and handle it */
- if (rtp->rtcp) {
+ if (rtcp) {
+ if (rtp->rtcp) {
+ }
return &ast_null_frame;
}
More information about the svn-commits
mailing list